Overview

A listing of new capabilities in Asterisk 1.8

In Brief

Asterisk 1.8 introduces a number of new features since the previous 1.6.2 release. Highlights include:

Secure RTP (SRTP)
IPv6 Support for SIP
Connected Party Identification Support - COLP and CONP.
Calendaring Integration for CalDAV, iCal, Exchange or EWS calendars
A new call logging system, Channel Event Logging (CEL)
Distributed Device State, including Message Waiting Indicator using Jabber/XMPP PubSub
Call Completion Supplementary Services (CCSS) Support, including Call Completion on Busy Subscriber (CCBS) and Call Completion on No Response (CCNR)
Advice of Charge, including AOC-S, AOC-D, and AOC-E Support
Multicast RTP
ISDN Q.SIG Call Rerouting and Call Deflection
Google Talk and Google Voice integration
Audio Pitch Shifting (for fun and profit)

Detailed Listing

SIP Changes

Added preferred_codec_only option in sip.conf. This feature limits the joint
codecs sent in response to an INVITE to the single most preferred codec.
Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
to be used for the outgoing call. It must be one of the codecs configured
for the device.
Added tlsprivatekey option to sip.conf. This allows a separate .pem file
to be used for holding a private key. If tlsprivatekey is not specified,
tlscertfile is searched for both public and private key.
Added tlsclientmethod option to sip.conf. This allows the protocol for
outbound client connections to be specified.
The sendrpid parameter has been expanded to include the options
'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
header to be sent (equivalent to setting sendrpid=yes) and setting
sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
The 'ignoresdpversion' behavior has been made automatic when the SDP received
is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
since the call will fail if Asterisk does not process the incoming SDP, Asterisk
will accept the SDP even if the SDP version number is not properly incremented,
but will generate a warning in the log indicating that the SIP peer that sent
the SDP should have the 'ignoresdpversion' option set.
The 'nat' option has now been been changed to have yes, no, force_rport, and
comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
remote side requests it and disables symmetric RTP support. Setting it to
force_rport forces RFC 3581 behavior and disables symmetric RTP support.
Setting it to comedia enables RFC 3581 behavior if the remote side requests it
and enables symmetric RTP support.
Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
response. This permits the master channel to know how each channel dialled
in a multi-channel setup resolved in an individual way.
Added 'externtcpport' and 'externtlsport' options to allow custom port
configuration for the externip and externhost options when tcp or tls is used.
Added support for message body (stored in content variable) to SIP NOTIFY message
accessible via AMI and CLI.
Added 'media_address' configuration option which can be used to explicitly specify
the IP address to use in the SDP for media (audio, video, and text) streams.
Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
received.
Added 'use_q850_reason' configuration option for generating and parsing
if available Reason: Q.850;cause=<cause code> header. It is implemented
in some gateways for better passing PRI/SS7 cause codes via SIP.
When dialing SIP peers, a new component may be added to the end of the dialstring
to indicate that a specific remote IP address or host should be used when dialing
the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
ability to selectively force bridged channels to also be encrypted is also
implemented. Branching in the dialplan can be done based on whether or not
a channel has secure media and/or signaling.
Added directmediapermit/directmediadeny to limit which peers can send direct media
to each other
Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
Charge messages to snom phones.
Added support for G.719 media streams.
Added support for 16khz signed linear media streams.
SIP is now able to bind to and communicate with IPv6 addresses. In addition,
RTP has been outfitted with the same abilities.
Added support for setting the Max-Forwards: header in SIP requests. Setting is
available in device configurations as well as in the dial plan.
Addition of the 'subscribe_network_change' option for turning on and off
res_stun_monitor module support in chan_sip.
Addition of the 'auth_options_requests' option for turning on and off
authentication for OPTIONS requests in chan_sip.

IAX2 Changes

Added rtsavesysname option into iax.conf to allow the systname to be saved
on realtime updates.
Added the ability for chan_iax2 to inform the dialplan whether or not
encryption is being used. This interoperates with the SIP SRTP implementation
so that a secure SIP call can be bridged to a secure IAX call when the
dialplan requires bridged channels to be "secure".
Addition of the 'subscribe_network_change' option for turning on and off
res_stun_monitor module support in chan_iax.

MGCP Changes

Added ability to preset channel variables on indicated lines with the setvar
configuration option. Also, clearvars=all resets the list of variables back
to none.
PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
See configs/res_pktccops.conf for more information.

XMPP Google Talk/Jingle changes

Added the externip option to gtalk.conf.
Added the stunaddr option to gtalk.conf which allows for the automatic
retrieval of the external ip from a stun server.

Applications

Added 'p' option to PickupChan() to allow for picking up channel by the first
match to a partial channel name.
Added .m3u support for Mp3Player application.
Added progress option to the app_dial D() option. When progress DTMF is
present, those values are sent immediately upon receiving a PROGRESS message
regardless if the call has been answered or not.
Added functionality to the app_dial F() option to continue with execution
at the current location when no parameters are provided.
Added the 'a' option to app_dial to answer the calling channel before any
announcements or macros are executed.
Modified app_dial to set answertime when the called channel answers even if
the called channel hangs up during playback of an announcement.
Modified app_dial 'r' option to support an additional parameter to play an
indication tone from indications.conf
Added c() option to app_chanspy. This option allows custom DTMF to be set
to cycle through the next available channel. By default this is still '*'.
Added x() option to app_chanspy. This option allows DTMF to be set to
exit the application.
The Voicemail application has been improved to automatically ignore messages
that only contain silence.
If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
associated mailbox(es) to be greetings-only.
The ChanSpy application now has the 'S' option, which makes the application
automatically exit once it hits a point where no more channels are available
to spy on.
The ChanSpy application also now has the 'E' option, which spies on a single
channel and exits when that channel hangs up.
The MeetMe application now turns on the DENOISE() function by default, for
each participant. In our tests, this has significantly decreased background
noise (especially noisy data centers).
Voicemail now permits storage of secrets in a separate file, located in the
spool directory of each individual user. The control for this is located in
the "passwordlocation" option in voicemail.conf. Please see the sample
configuration for more information.
The ChanIsAvail application now exposes the returned cause code using a separate
variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
Added 'd' option to app_followme. This option disables the "Please hold"
announcement.
Added 'y' option to app_record. This option enables a mode where any DTMF digit
received will terminate recording.
Voicemail now supports per mailbox settings for folders when using IMAP storage.
Previously the folder could only be set per context, but has now been extended
using the imapfolder option.
Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
Voicemail now allows the pager date format to be specified separately from the
email date format.
New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
to allow joining, leaving, and sending text to group chats.
MeetMe has a new option 'G' to play an announcement before joining a conference.
Page has a new option 'A(x)' which will playback an announcement simultaneously
to all paged phones (and optionally excluding the caller's one using the new
option 'n') before the call is bridged.
The 'f' option to Dial has been augmented to take an optional argument. If no
argument is provided, the 'f' option works as it always has. If an argument is
provided, then the connected party information of all outgoing channels created
during the Dial will be set to the argument passed to the 'f' option.
Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
Gosub on the peer.
The OSP lookup application adds in/outbound network ID, optional security,
number portability, QoS reporting, destination IP port, custom info and service
type features.
Added new application VMSayName that will play the recorded name of the voicemail
user if it exists, otherwise will play the mailbox number.
Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
retrieve state for a particular bridge, where <name> is the conference name
app_directory now allows exiting at any time using the operator or pound key.
Voicemail now supports setting a locale per-mailbox.
Two new applications are provided for declining counting phrases in multiple
languages. See the application notes for SayCountedNoun and SayCountedAdj for
more information.
Voicemail now runs the externnotify script when pollmailboxes is activated and
notices a change.
Voicemail now includes rdnis within msgXXXX.txt file.
Added 'D' command to ExternalIVR full details in http://wiki.asterisk.org

Dialplan Functions

SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
over SRV records associated with a specific service. From the CLI, type
'core show function SRVQUERY' and 'core show function SRVRESULT' for more
details on how these may be used.
PITCH_SHIFT dialplan function added. This function can be used to modify the
pitch of a channel's tx and rx audio streams.
Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
setting various connected line and redirecting party information.
CALLERID and CONNECTEDLINE dialplan functions have been extended to
support ISDN subaddressing.
The CHANNEL() function now supports the "name" and "checkhangup" options.
For DAHDI channels, the CHANNEL() dialplan function now allows
the dialplan to request changes in the configuration of the active
echo canceller on the channel (if any), for the current call only.
The syntax is:

exten => s,n,Set(CHANNEL(echocan_mode)=off)

The possible values are:

on - normal mode (the echo canceller is actually reinitialized)
off - disabled
fax - FAX/data mode (NLP disabled if possible, otherwise completely
disabled)
voice - voice mode (returns from FAX mode, reverting the changes that
were made when FAX mode was requested)

Added new dialplan function MASTER_CHANNEL(), which permits retrieving
and setting variables on the channel which created the current channel.
Administrators should take care to avoid naming conflicts, when multiple
channels are dialled at once, especially when used with the Local channel
construct (which all could set variables on the master channel). Usage
of the HASH() dialplan function, with the key set to the name of the slave
channel, is one approach that will avoid conflicts.
Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
audio in a channel.
func_odbc now allows multiple row results to be retrieved without using
mode=multirow. If rowlimit is set, then additional rows may be retrieved
from the same query by using the name of the function which retrieved the
first row as an argument to ODBC_FETCH().
Added JABBER_RECEIVE, which permits receiving XMPP messages from the
dialplan. This function returns the content of the received message.
Added REPLACE, which searches a given variable name for a set of characters,
then either replaces them with a single character or deletes them.
Added PASSTHRU, which literally passes the same argument back as its return
value. The intent is to be able to use a literal string argument to
functions that currently require a variable name as an argument.
HASH-associated variables now can be inherited across channel creation, by
prefixing the name of the hash at assignment with the appropriate number of
underscores, just like variables.
GROUP_MATCH_COUNT has been improved to allow regex matching on category
CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
whether or not channels that are bridged to the current channel will be
required to have secure signaling and/or media.
CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
the current channel has secure signaling and/or media.
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
Returns "0" if there is a B channel associated with the call.
Returns "1" if no B channel is associated with the call. The call is either
on hold or is a call waiting call.
Added option to dialplan function CDR(), the 'f' option
allows for high resolution times for billsec and duration fields.
FILE() now supports line-mode and writing.
Added FIELDNUM(), which returns the 1-based offset of a field in a list.
FRAME_TRACE(), for tracking internal ast_frames on a channel.

Dialplan Variables

Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
and is set when a dynamic feature is triggered.
Added PARKINGLOT which can be used with parkeddynamic feature.conf option
to dynamically create a new parking lot matching the value this varible is
set to.
Added PARKINGDYNAMIC which represents the template parkinglot defined in
features.conf that should be the base for dynamic parkinglots.
Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
parkinglot should have.
Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
should have.

Queue changes

Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
timeout has expired.
Added 'R' option to app_queue. This option stops moh and indicates ringing
to the caller when an Agent's phone is ringing. This can be used to indicate
to the caller that their call is about to be picked up, which is nice when
one has been on hold for an extened period of time.
A new config option, penaltymemberslimit, has been added to queues.conf.
When set this option will disregard penalty settings when a queue has too
few members.
A new option, 'I' has been added to both app_queue and app_dial.
By setting this option, Asterisk will not update the caller with
connected line changes or redirecting party changes when they occur.
A 'relative-peroidic-announce' option has been added to queues.conf. When
enabled, this option will cause periodic announce times to be calculated
from the end of announcements rather than from the beginning.
The autopause option in queues.conf can be passed a new value, "all." The
result is that if a member becomes auto-paused, he will be paused in all
queues for which he is a member, not just the queue that failed to reach
the member.
Added dialplan function QUEUE_EXISTS to check if a queue exists
The queue logger now allows events to optionally propagate to a file,
even when realtime logging is turned on. Additionally, realtime logging
supports sending the event arguments to 5 individual fields, although it
will fallback to the previous data definition, if the new table layout is
not found.

mISDN channel driver (chan_misdn) changes

Added display_connected parameter to misdn.conf to put a display string
in the CONNECT message containing the connected name and/or number if
the presentation setting permits it.
Added display_setup parameter to misdn.conf to put a display string
in the SETUP message containing the caller name and/or number if the
presentation setting permits it.
Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
indicate the dialplan settings are to be obtained from the asterisk
channel.
Made misdn.conf parameter callerid accept the "name" <number> format
used by the rest of the system.
Made use the nationalprefix and internationalprefix misdn.conf
parameters to prefix any received number from the ISDN link if that
number has the corresponding Type-Of-Number. NOTE: This includes
comparing the incoming call's dialed number against the MSN list.
Added the following new parameters: unknownprefix, netspecificprefix,
subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
received number from the ISDN link if that number has the corresponding
Type-Of-Number.
Added new dialplan application misdn_command which permits controlling
the CCBS/CCNR functionality.
Added new dialplan function mISDN_CC which permits retrieval of various
values from an active call completion record.
For PTP, you should manually send the COLR of the redirected-to party
for an incomming redirected call if the incoming call could experience
further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
set the REDIRECTING(to-pres) to the COLR. A call has been redirected
if the REDIRECTING(from-num) is not empty.
For outgoing PTP redirected calls, you now need to use the inhibit(i)
option on all of the REDIRECTING statements before dialing the
redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
and the REDIRECTING(from-xxx,i) values. The PTP call will update the
redirecting-to presentation (COLR) when it becomes available.
Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
information.

thirdparty mISDN enhancements

mISDN has been modified by Digium, Inc. to greatly expand facility message
support to allow:

Enhanced COLP support for call diversion and transfer.
CCBS/CCNR support.

The latest modified mISDN v1.1.x based version is available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk http://svn.digium.com/svn/thirdparty/mISDNuser/trunk

Tagged versions of the modified mISDN code are available under: http://svn.digium.com/svn/thirdparty/mISDN/tags http://svn.digium.com/svn/thirdparty/mISDNuser/tags

libpri channel driver (chan_dahdi) DAHDI changes

The channel variable PRIREDIRECTREASON is now just a status variable
and it is also deprecated. Use the REDIRECTING(reason) dialplan function
to read and alter the reason.
For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
redirected-to party for an incomming redirected call if the incoming call
could experience further redirects. Just set the
REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
to the COLR. A call has been redirected if the REDIRECTING(count) is not
zero.
For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
use the inhibit(i) option on all of the REDIRECTING statements before
dialing the redirected-to party. You still have to set the
REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
will update the redirecting-to presentation (COLR) when it becomes available.
Added the ability to ignore calls that are not in a Multiple Subscriber
Number (MSN) list for PTMP CPE interfaces.
Added dynamic range compression support for dahdi channels. It is
configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
Added support for ISDN calling and called subaddress with partial support
for connected line subaddress.
Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
Added standard location to add options to chan_dahdi dialing:
Dial(DAHDI/g1[/extension[/options]])
Current options:
K(<keypad_digits>)
R Reverse charging indication
Added Reverse Charging Indication (Collect calls) send/receive option.
Send reverse charging in SETUP message with the chan_dahdi R dialing option.
Dial(DAHDI/g1/extension/R)
Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
(requires latest LibPRI)
Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
dialing option. Dial(DAHDI/g1/[~mdavenport:extension]/K(<keypad_digits>))
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
(requires latest LibPRI)
Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
to eliminate tromboned calls. A tromboned call goes out an interface and comes
back into the same interface. Tromboned calls happen because of call routing,
call deflection, call forwarding, and call transfer.
Added the ability to send and receive ETSI Advice-Of-Charge messages.
Added the ability to support call waiting calls. (The SETUP has no B channel
assigned.)
Added Malicious Call ID (MCID) event to the AMI call event class.
Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).

Asterisk Manager Interface

The Hangup action now accepts a Cause header which may be used to
set the channel's hangup cause.
sslprivatekey option added to manager.conf and http.conf. Adds the ability
to specify a separate .pem file to hold a private key. By default sslcert
is used to hold both the public and private key.
Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
for options containing the 'tls' prefix. For example, 'sslenable' is now
'tlsenable'. This has been done in effort to keep ssl and tls options consistent
across all .conf files. All affected sample.conf files have been modified to
reflect this change. Previous options such as 'sslenable' still work,
but options with the 'tls' prefix are preferred.
Added a MuteAudio AMI action for muting inbound and/or outbound audio
in a channel. (res_mutestream.so)
The configuration file manager.conf now supports a channelvars option, which
specifies a list of channel variables to include in each channel-oriented
event.
The redirect command now has new parameters ExtraContext, ExtraExtension,
and ExtraPriority to allow redirecting the second channel to a different
location than the first.
Added new event "JabberStatus" in the Jabber module to monitor buddies
status.
Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
in a MixMonitor recording.
The 'iax2 show peers' output is now similar to the expected output of
'sip show peers'.
Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
aoc event class.
Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
AOC-E messages on a channel.
A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
conform more closely to similar events.
Added a new eventfilter option per user to allow whitelisting and blacklisting
of events.
Added optional parkinglot variable for park command.

Channel Event Logging

A new interface, CEL, is introduced here. CEL logs single events, much like
the AMI, but it differs from the AMI in that it logs to db backends much
like CDR does; is based on the event subsystem introduced by Russell, and
can share in all its benefits; allows multiple backends to operate like CDR;
is specialized to event data that would be of concern to billing sytems,
like CDR. Backends for logging and accounting calls have been produced,
but a new CDR backend is still in development.

CDR

'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
Multiple files and formats can now be specified in cdr_custom.conf.
cdr_syslog has been added which allows CDRs to be written directly to syslog.
See configs/cdr_syslog.conf.sample for more information.
A 'sequence' field has been added to CDRs which can be combined with
linkedid or uniqueid to uniquely identify a CDR.
Handling of billsec and duration field has changed. If your table definition
specifies those fields as float,double or similar they will now be logged with
microsecond accuracy instead of a whole integer.

Calendaring for Asterisk

A new set of modules were added supporing calendar integration with Asterisk.
Dialplan functions for reading from and writing to calendars are included,
as well as the ability to execute dialplan logic upon calendar event notifications.
iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2003 support does not support forms-based authentication).

Call Completion Supplementary Services for Asterisk

Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
DAHDI/ISDN supports call completion for the following switch types:
EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
See http://wiki.asterisk.org for details.

Multicast RTP Support

A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP
paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
Type can be either basic or linksys.
Destination is the IP address and port for the RTP packets.
Control address is specific to the linksys type and is used for sending the control
packets unique to them.

Security Events Framework

Asterisk has a new C API for reporting security events. The module res_security_log
sends these events to the "security" logger level. Currently, AMI is the only
Asterisk component that reports security events. However, SIP support will be
coming soon. For more information on the security events framework, see the
"Security Events" chapter of the included documentation - doc/AST.pdf.

Fax

A technology independent fax frontend (res_fax) has been added to Asterisk.
A spandsp based fax backend (res_fax_spandsp) has been added.
The app_fax module has been deprecated in favor of the res_fax module and
the new res_fax_spandsp backend.
The SendFAX and ReceiveFAX applications now send their log messages to a
'fax' logger level, instead of to the generic logger levels. To see these
messages, the system's logger.conf file will need to direct the 'fax' logger
level to one or more destinations; the logger.conf.sample file includes an
example of how to do this. Note that if the 'fax' logger level is not
directed to at least one destination, log messages generated by these
applications will be lost, and that if the 'fax' logger level is directed to
the console, the 'core set verbose' and 'core set debug' CLI commands will
have no effect on whether the messages appear on the console or not.

Miscellaneous

The transmit_silence_during_record option in asterisk.conf.sample has been removed.
Now, in order to enable transmitting silence during record the transmit_silence
option should be used. transmit_silence_during_record remains a valid option, but
defaults to the behavior of the transmit_silence option.
Addition of the Unit Test Framework API for managing registration and execution
of unit tests with the purpose of verifying the operation of C functions.
SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
XMPP text messages to the remote JID.
Modules.conf has a new option - "require" - that marks a module as critical for
the execution of Asterisk.
If one of the required modules fail to load, Asterisk will exit with a return
code set to 2.
An 'X' option has been added to the asterisk application which enables #exec support.
This allows #exec to be used in asterisk.conf.
jabber.conf supports a new option auth_policy that toggles auto user registration.
A new lockconfdir option has been added to asterisk.conf to protect the
configuration directory (/etc/asterisk by default) during reloads.
The parkeddynamic option has been added to features.conf to enable the creation
of dynamic parkinglots.
chan_dahdi now supports reporting alarms over AMI either by channel or span via
the reportalarms config option.
chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
The Realtime dialplan switch now caches entries for 1 second. This provides a
significant increase in performance (about 3X) for installations using this switchtype.
Distributed devicestate now supports the use of the XMPP protocol, in addition to
AIS. For more information, please see http://wiki.asterisk.org
The addition of G.719 pass-through support.
Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
during device configuration.
The UNISTIM channel driver (chan_unistim) has been updated to support devices that
have less than 3 lines on the LCD.
Realtime now supports database failover. See the sample extconfig.conf for details.
The addition of improved translation path building for wideband codecs. Sample
rate changes during translation are now avoided unless absolutely necessary.
The addition of the res_stun_monitor module for monitoring and reacting to network
changes while behind a NAT.

CLI Changes

The 'core set debug' and 'core set verbose' commands, in previous versions, could
optionally accept a filename, to apply the setting only to the code generated from
that source file when Asterisk was built. However, there are some modules in Asterisk
that are composed of multiple source files, so this did not result in the behavior
that users expected. In this version, 'core set debug' and 'core set verbose'
can optionally accept module names instead (with or without the .so extension),
which applies the setting to the entire module specified, regardless of which source
files it was built from.
New 'manager show settings' command showing the current settings loaded from
manager.conf.
Added 'all' keyword to the CLI command "channel request hangup" so that you can send
the channel hangup request to all channels.
Added a "core reload" CLI command that executes a global reload of Asterisk.

New in 1.8